U.S. patent application number 14/101777 was filed with the patent office on 2015-06-11 for systems and methods for providing adaptive playback equalization in an audio device.
This patent application is currently assigned to Cirrus Logic, Inc.. The applicant listed for this patent is Cirrus Logic, Inc.. Invention is credited to Jeffrey D. Alderson, Jon D. Hendrix.
Application Number | 20150161980 14/101777 |
Document ID | / |
Family ID | 51799347 |
Filed Date | 2015-06-11 |
United States Patent
Application |
20150161980 |
Kind Code |
A1 |
Alderson; Jeffrey D. ; et
al. |
June 11, 2015 |
SYSTEMS AND METHODS FOR PROVIDING ADAPTIVE PLAYBACK EQUALIZATION IN
AN AUDIO DEVICE
Abstract
In accordance with systems and methods of the present
disclosure, a method may include receiving an error microphone
signal indicative of an acoustic output of a transducer and ambient
audio sounds at the acoustic output of the transducer. The method
may also include generating an anti-noise signal to reduce the
presence of the ambient audio sounds at the acoustic output of the
transducer based at least on the error microphone signal. The
method may further include generating an equalized source audio
signal from a source audio signal by adapting, based at least on
the error microphone signal, a response of the adaptive playback
equalization system to minimize a difference between the source
audio signal and the error microphone signal. The method may
additionally include combining the anti-noise signal with the
equalized source audio signal to generate an audio signal provided
to the transducer.
Inventors: |
Alderson; Jeffrey D.;
(Austin, TX) ; Hendrix; Jon D.; (Wimberley,
TX) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
Cirrus Logic, Inc. |
Austin |
TX |
US |
|
|
Assignee: |
Cirrus Logic, Inc.
Austin
TX
|
Family ID: |
51799347 |
Appl. No.: |
14/101777 |
Filed: |
December 10, 2013 |
Current U.S.
Class: |
381/71.11 |
Current CPC
Class: |
G10K 11/17817 20180101;
G10K 11/17885 20180101; G10K 2210/1081 20130101; H04R 2410/05
20130101; G10K 11/17881 20180101; H04R 3/04 20130101; G10K 11/17854
20180101 |
International
Class: |
G10K 11/175 20060101
G10K011/175 |
Claims
1. A personal audio device comprising: a personal audio device
housing; a transducer coupled to the housing for reproducing an
output audio signal including an equalized source audio signal for
playback to a listener and an anti-noise signal for countering the
effects of ambient audio sounds in an acoustic output of the
transducer; an error microphone coupled to the housing in proximity
to the transducer for providing an error microphone signal
indicative of the acoustic output of the transducer and the ambient
audio sounds at the transducer; one or more processing circuits
that implement: a noise cancellation system that generates the
anti-noise signal to reduce the presence of the ambient audio
sounds heard by the listener based at least on the error microphone
signal; and an adaptive playback equalization system that generates
the equalized source audio signal from a source audio signal by
adapting, based at least on the error microphone signal, a response
of the adaptive playback equalization system to minimize a
difference between the source audio signal and the error microphone
signal.
2. The personal audio device of claim 1, wherein the adaptive
playback equalization system comprises: an adaptive equalization
filter having a response that generates the equalized source audio
signal from the source audio signal to reduce the effects of an
electro-acoustical path of the source audio signal through the
transducer; and a coefficient control block that shapes the
response of the adaptive equalization filter in conformity with the
error microphone signal and the source audio signal by adapting the
response of the adaptive equalization filter to minimize the
difference between the error microphone signal and the source audio
signal.
3. The personal audio device of claim 2, wherein the adaptive
equalization filter comprises a shelving filter, wherein at least
one of a pole frequency and a zero frequency of the shelving filter
are variable based on the error microphone signal.
4. The personal audio device of claim 2, wherein the adaptive
playback equalization system further comprises a secondary path
estimate filter for modeling the electro-acoustical path and having
a response that generates a secondary path estimate from the source
audio signal and wherein the coefficient control block shapes the
response of the adaptive equalization filter in conformity with the
secondary path estimate and a delay corrected error, wherein the
delay corrected error is based on a difference between the error
microphone signal and a delayed source audio signal.
5. The personal audio device of claim 4, wherein the one or more
processing circuits implement a second coefficient control block
that shapes the response of the secondary path estimate filter in
conformity with the source audio signal and a playback corrected
error by adapting the response of the secondary path estimate
filter to minimize the playback corrected error, wherein the
playback corrected error is based on a difference between the error
microphone signal and the secondary path estimate.
6. The personal audio device of claim 4, wherein a number of
coefficients of the coefficient control block is selected such that
a magnitude of the response of the adaptive equalization filter
corresponding to a frequency in which the response of the secondary
path estimate filter is substantially zero is limited below a
predetermined maximum.
7. The personal audio device of claim 4, wherein the one or more
processing circuits implement a noise injection portion for
injecting respective noise signals into the secondary path estimate
and the delay corrected error in order to bias, to below a
predetermined maximum, a magnitude of the response of the adaptive
equalization filter corresponding to a frequency in which the
response of the secondary path estimate filter is substantially
zero.
8. The personal audio device of claim 1, wherein the one or more
processing circuits disable the response of the adaptive playback
equalization system from adapting responsive to at least one of: a
determination that a spectral density of the source audio signal is
lesser than a minimum spectral density; a determination that the
transducer has been removed from a proximity of an ear of the
listener; a determination that a magnitude of the output audio
signal is within a predetermined threshold of a magnitude of a
power supply for driving the output audio signal; and a
determination that a displacement of the transducer is such that
its displacement as a function of the output audio signal is
substantially nonlinear.
9. The personal audio device of claim 1, further comprising a
reference microphone coupled to the housing for providing a
reference microphone signal indicative of the ambient audio sounds,
wherein the noise cancellation system further comprises: an
adaptive filter having a response that generates the anti-noise
signal from the reference microphone signal to reduce the presence
of the ambient audio sounds heard by the listener; and a
coefficient control block that shapes the response of the adaptive
filter in conformity with the error microphone signal and the
reference microphone signal by adapting the response of the
adaptive filter to minimize the ambient audio sounds in the error
microphone signal.
10. The personal audio device of claim 1, further comprising a
reference microphone coupled to the housing for providing a
reference microphone signal indicative of the ambient audio sounds,
wherein the noise cancellation system further comprises: a filter
having a response that generates the anti-noise signal from the
reference microphone signal to reduce the presence of the ambient
audio sounds heard by the listener; a secondary path estimate
adaptive filter for modeling an electro-acoustical path of the
source audio signal and having a response that generates a
secondary path estimate from the equalized source audio signal; and
a coefficient control block that shapes the response of the
secondary path estimate adaptive filter in conformity with the
equalized source audio signal and a playback corrected error by
adapting the response of the secondary path estimate filter to
minimize the playback corrected error, wherein the playback
corrected error is based on a difference between the error
microphone signal and the secondary path estimate.
11. The personal audio device of claim 10, wherein the one or more
processing circuits are configured to adapt the response of the
secondary path estimate adaptive filter prior to adapting the
response of the adaptive playback equalization system.
12. The personal audio device of claim 11, wherein the one or more
processing circuits are configured to alternate adaptation of the
secondary path estimate adaptive filter and the response of the
adaptive playback equalization system.
13. The personal audio device of claim 10, wherein the one or more
processing circuits are configured to adapt the response of the
adaptive playback equalization system only when the secondary path
estimate adaptive filter is adapting.
14. The personal audio device of claim 10, wherein the one or more
processing circuits are configured to adapt the response of the
adaptive playback equalization system at a rate slower than the
rate of adaptation of the secondary path estimate adaptive
filter.
15. A method comprising: receiving an error microphone signal
indicative of an acoustic output of a transducer and ambient audio
sounds at the acoustic output of the transducer; generating an
anti-noise signal to reduce the presence of the ambient audio
sounds at the acoustic output of the transducer based at least on
the error microphone signal; generating an equalized source audio
signal from a source audio signal by adapting, based at least on
the error microphone signal, a response of an adaptive playback
equalization system to minimize a difference between the source
audio signal and the error microphone signal; and combining the
anti-noise signal with the equalized source audio signal to
generate an audio signal provided to the transducer.
16. The method of claim 15, wherein the equalized source audio
signal is generated by an adaptive equalization filter having a
response that generates the equalized source audio signal from the
source audio signal to reduce the effects of an electro-acoustical
path of the source audio signal through the transducer, and the
method further comprising shaping the response of the adaptive
equalization filter in conformity with the error microphone signal
and the source audio signal by adapting the response of the
adaptive equalization filter to minimize the difference between the
error microphone signal and the source audio signal.
17. The method of claim 16, wherein the adaptive equalization
filter comprises a shelving filter, wherein at least one of a pole
frequency and a zero frequency of the shelving filter are variable
based on the error microphone signal.
18. The method of claim 16, further comprising generating a
secondary path estimate from the source audio signal by filtering
the source audio signal with a secondary path estimate filter
modeling an electro-acoustical path of the source audio signal; and
wherein shaping the response of the adaptive equalization filter
comprises shaping the response of the adaptive equalization filter
in conformity with the secondary path estimate and a delay
corrected error, wherein the delay corrected error is based on a
difference between the error microphone signal and a delayed source
audio signal.
19. The method of claim 18, further comprising shaping the response
of the secondary path estimate filter in conformity with the source
audio signal and a playback corrected error by adapting the
response of the secondary path estimate filter to minimize the
playback corrected error, wherein the playback corrected error is
based on a difference between the error microphone signal and the
secondary path estimate.
20. The method of claim 18, wherein the response of the adaptive
equalization filter is shaped by a coefficient control block, and a
number of coefficients of the coefficient control block is selected
such that a magnitude of the response of the adaptive equalization
filter corresponding to a frequency in which the response of the
secondary path estimate filter is substantially zero is limited
below a predetermined maximum.
21. The method of claim 18, further comprising injecting respective
noise signals into the secondary path estimate and the delay
corrected error in order to bias, to below a predetermined maximum,
a magnitude of the response of the adaptive equalization filter
corresponding to a frequency in which the response of the secondary
path estimate filter is substantially zero.
22. The method of claim 15, further comprising disabling the
response of the adaptive playback equalization system from adapting
responsive to at least one of: a determination that a spectral
density of the source audio signal is lesser than a minimum
spectral density; a determination that the transducer has been
removed from a proximity of an ear of the listener; a determination
that a magnitude of the output audio signal is within a
predetermined threshold of a magnitude of a power supply for
driving the output audio signal; and a determination that a
displacement of the transducer is such that its displacement as a
function of the output audio signal is substantially nonlinear.
23. The method of claim 15, further comprising: receiving a
reference microphone signal indicative of the ambient audio sounds;
and generating the anti-noise signal from filtering the reference
microphone signal with an adaptive filter to reduce the presence of
the ambient audio sounds heard by the listener by shaping the
response of the adaptive filter in conformity with the error
microphone signal and the reference microphone signal by adapting
the response of the adaptive filter to minimize the ambient audio
sounds in the error microphone signal.
24. The method of claim 15, further comprising: receiving a
reference microphone signal indicative of the ambient audio sounds;
generating the anti-noise signal from the reference microphone
signal to reduce the presence of the ambient audio sounds heard by
the listener; generating a secondary path estimate from the
equalized source audio signal by filtering the equalized source
audio signal with a secondary path estimate filter modeling an
electro-acoustical path of the source audio signal; and shaping the
response of the secondary path estimate filter in conformity with
the equalized source audio signal and a playback corrected error by
adapting the response of the secondary path estimate filter to
minimize the playback corrected error, wherein the playback
corrected error is based on a difference between the error
microphone signal and the secondary path estimate.
25. The method of claim 24, wherein the response of the secondary
path estimate adaptive filter adapts prior to adaptation of the
response of the adaptive playback equalization system.
26. The method of claim 25, further comprising alternating
adaptation of the secondary path estimate adaptive filter and the
response of the adaptive playback equalization system.
27. The method of claim 24, further comprising adapting the
response of the adaptive playback equalization system only when the
secondary path estimate adaptive filter is adapting.
28. The method claim 24, further comprising adapting the response
of the adaptive playback equalization system at a rate slower than
the rate of adaptation of the secondary path estimate adaptive
filter.
29. An integrated circuit for implementing at least a portion of a
personal audio device, comprising: an output for providing a signal
to a transducer including both an equalized source audio signal for
playback to a listener and an anti-noise signal for countering the
effect of ambient audio sounds in an acoustic output of the
transducer; an error microphone input for receiving an error
microphone signal indicative of the acoustic output of the
transducer and the ambient audio sounds at the transducer; and one
or more processing circuits that implement: a noise cancellation
system that generates the anti-noise signal to reduce the presence
of the ambient audio sounds heard by the listener based at least on
the error microphone signal; and an adaptive playback equalization
system that generates the equalized source audio signal from a
source audio signal by adapting, based at least on the error
microphone signal, a response of the adaptive playback equalization
system to minimize a difference between the source audio signal and
the error microphone signal.
30. The integrated circuit of claim 29, wherein the adaptive
playback equalization system comprises: an adaptive equalization
filter having a response that generates the equalized source audio
signal from the source audio signal to reduce the effects of an
electro-acoustical path of the source audio signal through the
transducer; and a coefficient control block that shapes the
response of the adaptive equalization filter in conformity with the
error microphone signal and the source audio signal by adapting the
response of the adaptive equalization filter to minimize the
difference between the error microphone signal and the source audio
signal.
31. The integrated circuit of claim 30, wherein the adaptive
equalization filter comprises a shelving filter, wherein at least
one of a pole frequency and a zero frequency of the shelving filter
are variable based on the error microphone signal.
32. The integrated circuit of claim 30, wherein the adaptive
playback equalization system further comprises a secondary path
estimate filter for modeling the electro-acoustical path and having
a response that generates a secondary path estimate from the source
audio signal and wherein the coefficient control block shapes the
response of the adaptive equalization filter in conformity with the
secondary path estimate and a delay corrected error, wherein the
delay corrected error is based on a difference between the error
microphone signal and a delayed source audio signal.
33. The integrated circuit of claim 32, wherein the one or more
processing circuits implement a second coefficient control block
that shapes the response of the secondary path estimate filter in
conformity with the source audio signal and a playback corrected
error by adapting the response of the secondary path estimate
filter to minimize the playback corrected error, wherein the
playback corrected error is based on a difference between the error
microphone signal and the secondary path estimate.
34. The integrated circuit of claim 32, wherein a number of
coefficients of the coefficient control block is selected such that
a magnitude of the response of the adaptive equalization filter
corresponding to a frequency in which the response of the secondary
path estimate filter is substantially zero is limited below a
predetermined maximum.
35. The integrated circuit of claim 32, wherein the one or more
processing circuits implement a noise injection portion for
injecting respective noise signals into the secondary path estimate
and the delay corrected error in order to bias, to below a
predetermined maximum, a magnitude of the response of the adaptive
equalization filter corresponding to a frequency in which the
response of the secondary path estimate filter is substantially
zero.
36. The integrated circuit of claim 29, wherein the one or more
processing circuits disable the response of the adaptive playback
equalization system from adapting responsive to at least one of: a
determination that a spectral density of the source audio signal is
lesser than a minimum spectral density; a determination that the
transducer has been removed from a proximity of an ear of the
listener; a determination that a magnitude of the output audio
signal is within a predetermined threshold of a magnitude of a
power supply for driving the output audio signal; and a
determination that a displacement of the transducer is such that
its displacement as a function of the output audio signal is
substantially nonlinear.
37. The integrated circuit of claim 29, further comprising a
reference microphone input for receiving a reference microphone
signal indicative of the ambient audio sounds, wherein the noise
cancellation system further comprises: an adaptive filter having a
response that generates the anti-noise signal from the reference
microphone signal to reduce the presence of the ambient audio
sounds heard by the listener; and a coefficient control block that
shapes the response of the adaptive filter in conformity with the
error microphone signal and the reference microphone signal by
adapting the response of the adaptive filter to minimize the
ambient audio sounds in the error microphone signal.
38. The integrated circuit of claim 29, further comprising a
reference microphone input for receiving a reference microphone
signal indicative of the ambient audio sounds, wherein the noise
cancellation system further comprises: a filter having a response
that generates the anti-noise signal from the reference microphone
signal to reduce the presence of the ambient audio sounds heard by
the listener; a secondary path estimate adaptive filter for
modeling an electro-acoustical path of the source audio signal and
having a response that generates a secondary path estimate from the
equalized source audio signal; and a coefficient control block that
shapes the response of the secondary path estimate adaptive filter
in conformity with the equalized source audio signal and a playback
corrected error by adapting the response of the secondary path
estimate adaptive filter to minimize the playback corrected error,
wherein the playback corrected error is based on a difference
between the error microphone signal and the secondary path
estimate.
39. The integrated circuit of claim 38, wherein the one or more
processing circuits are configured to adapt the response of the
secondary path estimate adaptive filter prior to adapting the
response of the adaptive playback equalization system.
40. The integrated circuit of claim 39, wherein the one or more
processing circuits are configured to alternate adaptation of the
secondary path estimate adaptive filter and the response of the
adaptive playback equalization system.
41. The integrated circuit of claim 38, wherein the one or more
processing circuits are configured to adapt the response of the
adaptive playback equalization system only when the secondary path
estimate adaptive filter is adapting.
42. The integrated circuit of claim 38, wherein the one or more
processing circuits are configured to adapt the response of the
adaptive playback equalization system at a rate slower than the
rate of adaptation of the secondary path estimate adaptive filter.
Description
FIELD OF DISCLOSURE
[0001] The present disclosure relates in general to adaptive noise
cancellation in connection with an acoustic transducer, and more
particularly, to providing for adaptive playback equalization in an
audio device.
BACKGROUND
[0002] Personal audio devices, such as mobile/cellular telephones,
cordless telephones, and other consumer audio devices, such as mp3
players, are in widespread use. Performance of such devices with
respect to intelligibility can be improved by providing noise
canceling using a microphone to measure ambient acoustic events and
then using signal processing to insert an anti-noise signal into
the output of the device to cancel the ambient acoustic events.
Because the acoustic environment around personal audio devices such
as wireless telephones can change dramatically, depending on the
sources of noise that are present and the position of the device
itself, it is desirable to adapt the noise canceling to take into
account such environmental changes.
[0003] Some personal audio devices also include equalizers.
Equalizers typically attempt to apply to a source audio signal an
inverse of a response of the electro-acoustic path of the source
audio signal through the transducer, in order to reduce the effects
of the electro-acoustic path. In most traditional approaches,
equalization is performed with a static equalizer. However, an
adaptive equalizer may provide better output sound quality than a
static equalizer, and thus, may be desirable in many
applications.
SUMMARY
[0004] In accordance with the teachings of the present disclosure,
the disadvantages and problems associated with improving audio
performance of a personal audio device may be reduced or
eliminated.
[0005] In accordance with embodiments of the present disclosure, a
personal audio device may include a personal audio device housing,
a transducer, an error microphone, and one or more processing
circuits. The transducer may be coupled to the housing for
reproducing an output audio signal including an equalized source
audio signal for playback to a listener and an anti-noise signal
for countering the effects of ambient audio sounds in an acoustic
output of the transducer. The error microphone may be coupled to
the housing in proximity to the transducer for providing an error
microphone signal indicative of the acoustic output of the
transducer and the ambient audio sounds at the transducer. The one
or more processing circuits may implement: a noise cancellation
system that generates the anti-noise signal to reduce the presence
of the ambient audio sounds heard by the listener based at least on
the error microphone signal and an adaptive playback equalization
system that generates the equalized source audio signal from a
source audio signal by adapting, based at least on the error
microphone signal, a response of the adaptive playback equalization
system to minimize a difference between the source audio signal and
the error microphone signal.
[0006] In accordance with these and other embodiments of the
present disclosure, a method may include receiving an error
microphone signal indicative of an acoustic output of a transducer
and ambient audio sounds at the acoustic output of the transducer.
The method may also include generating an anti-noise signal to
reduce the presence of the ambient audio sounds at the acoustic
output of the transducer based at least on the error microphone
signal. The method may further include generating an equalized
source audio signal from a source audio signal by adapting, based
at least on the error microphone signal, a response of the adaptive
playback equalization system to minimize a difference between the
source audio signal and the error microphone signal. The method may
additionally include combining the anti-noise signal with the
equalized source audio signal to generate an audio signal provided
to the transducer.
[0007] In accordance with these and other embodiments of the
present disclosure, an integrated circuit for implementing at least
a portion of a personal audio device may include an output, an
error microphone input, and one or more processing circuits. The
output may be configured to provide a signal to a transducer
including both an equalized source audio signal for playback to a
listener and an anti-noise signal for countering the effect of
ambient audio sounds in an acoustic output of the transducer. The
error microphone may be configured to receive an error microphone
signal indicative of the output of the transducer and the ambient
audio sounds at the transducer. The one or more processing circuits
may implement: a noise cancellation system that generates the
anti-noise signal to reduce the presence of the ambient audio
sounds heard by the listener based at least on the error microphone
signal and an adaptive playback equalization system that generates
the equalized source audio signal from a source audio signal by
adapting, based at least on the error microphone signal, a response
of the adaptive playback equalization system to minimize a
difference between the source audio signal and the error microphone
signal.
[0008] Technical advantages of the present disclosure may be
readily apparent to one of ordinary skill in the art from the
figures, description and claims included herein. The objects and
advantages of the embodiments will be realized and achieved at
least by the elements, features, and combinations particularly
pointed out in the claims.
[0009] It is to be understood that both the foregoing general
description and the following detailed description are examples and
explanatory and are not restrictive of the claims set forth in this
disclosure.
BRIEF DESCRIPTION OF THE DRAWINGS
[0010] A more complete understanding of the present embodiments and
advantages thereof may be acquired by referring to the following
description taken in conjunction with the accompanying drawings, in
which like reference numbers indicate like features, and
wherein:
[0011] FIG. 1A is an illustration of an example personal audio
device, in accordance with embodiments of the present
disclosure;
[0012] FIG. 1B is an illustration of an example personal audio
device with a headphone assembly coupled thereto, in accordance
with embodiments of the present disclosure;
[0013] FIG. 2 is a block diagram of selected circuits within the
personal audio device depicted in FIG. 1, in accordance with
embodiments of the present disclosure;
[0014] FIG. 3 is a block diagram depicting selected signal
processing circuits and functional blocks within an example active
noise canceling (ANC) circuit of a coder-decoder (CODEC) integrated
circuit of FIG. 3, in accordance with embodiments of the present
disclosure;
[0015] FIG. 4 is a block diagram depicting selected signal
processing circuits and functional blocks within an example
adaptive equalization circuit of a coder-decoder (CODEC) integrated
circuit of FIG. 3, in accordance with embodiments of the present
disclosure; and
[0016] FIG. 5 is a block diagram depicting selected signal
processing circuits and functional blocks within an example noise
injection portion of an adaptive equalization circuit of FIG. 4, in
accordance with embodiments of the present disclosure.
DETAILED DESCRIPTION
[0017] Referring now to FIG. 1A, a personal audio device 10 as
illustrated in accordance with embodiments of the present
disclosure is shown in proximity to a human ear 5. Personal audio
device 10 is an example of a device in which techniques in
accordance with embodiments of the invention may be employed, but
it is understood that not all of the elements or configurations
embodied in illustrated personal audio device 10, or in the
circuits depicted in subsequent illustrations, are required in
order to practice the invention recited in the claims. Personal
audio device 10 may include a transducer such as speaker SPKR that
reproduces distant speech received by personal audio device 10,
along with other local audio events such as ringtones, stored audio
program material, injection of near-end speech (i.e., the speech of
the user of personal audio device 10) to provide a balanced
conversational perception, and other audio that requires
reproduction by personal audio device 10, such as sources from
webpages or other network communications received by personal audio
device 10 and audio indications such as a low battery indication
and other system event notifications. A near-speech microphone NS
may be provided to capture near-end speech, which is transmitted
from personal audio device 10 to the other conversation
participant(s).
[0018] Personal audio device 10 may include adaptive noise
cancellation (ANC) circuits and features that inject an anti-noise
signal into speaker SPKR to improve intelligibility of the distant
speech and other audio reproduced by speaker SPKR. A reference
microphone R may be provided for measuring the ambient acoustic
environment, and may be positioned away from the typical position
of a user's mouth, so that the near-end speech may be minimized in
the signal produced by reference microphone R. Another microphone,
error microphone E, may be provided in order to further improve the
ANC operation by providing a measure of the ambient audio combined
with the audio reproduced by speaker SPKR close to ear 5, when
personal audio device 10 is in close proximity to ear 5. Circuit 14
within personal audio device 10 may include an audio CODEC
integrated circuit (IC) 20 that receives the signals from reference
microphone R, near-speech microphone NS, and error microphone E,
and interfaces with other integrated circuits such as a
radio-frequency (RF) integrated circuit 12 having a wireless
telephone transceiver. In some embodiments of the disclosure, the
circuits and techniques disclosed herein may be incorporated in a
single integrated circuit that includes control circuits and other
functionality for implementing the entirety of the personal audio
device, such as an MP3 player-on-a-chip integrated circuit. In
these and other embodiments, the circuits and techniques disclosed
herein may be implemented partially or fully in software and/or
firmware embodied in computer-readable media and executable by a
controller or other processing device.
[0019] In general, ANC techniques of the present disclosure measure
ambient acoustic events (as opposed to the output of speaker SPKR
and/or the near-end speech) impinging on reference microphone R,
and by also measuring the same ambient acoustic events impinging on
error microphone E, ANC processing circuits of personal audio
device 10 adapt an anti-noise signal generated out the output of
speaker SPKR from the output of reference microphone R to have a
characteristic that minimizes the amplitude of the ambient acoustic
events at error microphone E. Because acoustic path P(z) extends
from reference microphone R to error microphone E, ANC circuits are
effectively estimating acoustic path P(z) while removing effects of
an electro-acoustic path S(z) that represents the response of the
audio output circuits of CODEC IC 20 and the acoustic/electric
transfer function of speaker SPKR including the coupling between
speaker SPKR and error microphone E in the particular acoustic
environment, which may be affected by the proximity and structure
of ear 5 and other physical objects and human head structures that
may be in proximity to personal audio device 10, when personal
audio device 10 is not firmly pressed to ear 5. While the
illustrated personal audio device 10 includes a two-microphone ANC
system with a third near-speech microphone NS, some aspects of the
present invention may be practiced in a system that does not
include separate error and reference microphones, or a wireless
telephone that uses near-speech microphone NS to perform the
function of the reference microphone R. Also, in personal audio
devices designed only for audio playback, near-speech microphone NS
will generally not be included, and the near-speech signal paths in
the circuits described in further detail below may be omitted,
without changing the scope of the disclosure, other than to limit
the options provided for input to the microphone covering detection
schemes. In addition, although only one reference microphone R is
depicted in FIG. 1, the circuits and techniques herein disclosed
may be adapted, without changing the scope of the disclosure, to
personal audio devices including a plurality of reference
microphones.
[0020] Referring now to FIG. 1B, personal audio device 10 is
depicted having a headphone assembly 13 coupled to it via audio
port 15. Audio port 15 may be communicatively coupled to RF
integrated circuit 12 and/or CODEC IC 20, thus permitting
communication between components of headphone assembly 13 and one
or more of RF integrated circuit 12 and/or CODEC IC 20. As shown in
FIG. 1B, headphone assembly 13 may include a combox 16, a left
headphone 18A, and a right headphone 18B. As used in this
disclosure, the term "headphone" broadly includes any loudspeaker
and structure associated therewith that is intended to be
mechanically held in place proximate to a listener's ear or ear
canal, and includes without limitation earphones, earbuds, and
other similar devices. As more specific non-limiting examples,
"headphone," may refer to intra-canal earphones, intra-concha
earphones, supra-concha earphones, and supra-aural earphones.
[0021] Combox 16 or another portion of headphone assembly 13 may
have a near-speech microphone NS to capture near-end speech in
addition to or in lieu of near-speech microphone NS of personal
audio device 10. In addition, each headphone 18A, 18B may include a
transducer such as speaker SPKR that reproduces distant speech
received by personal audio device 10, along with other local audio
events such as ringtones, stored audio program material, injection
of near-end speech (i.e., the speech of the user of personal audio
device 10) to provide a balanced conversational perception, and
other audio that requires reproduction by personal audio device 10,
such as sources from webpages or other network communications
received by personal audio device 10 and audio indications such as
a low battery indication and other system event notifications. Each
headphone 18A, 18B may include a reference microphone R for
measuring the ambient acoustic environment and an error microphone
E for measuring of the ambient audio combined with the audio
reproduced by speaker SPKR close to a listener's ear when such
headphone 18A, 18B is engaged with the listener's ear. In some
embodiments, CODEC IC 20 may receive the signals from reference
microphone R, near-speech microphone NS, and error microphone E of
each headphone and perform adaptive noise cancellation for each
headphone as described herein. In other embodiments, a CODEC IC or
another circuit may be present within headphone assembly 13,
communicatively coupled to reference microphone R, near-speech
microphone NS, and error microphone E, and configured to perform
adaptive noise cancellation as described herein.
[0022] The various microphones referenced in this disclosure,
including reference microphones, error microphones, and near-speech
microphones, may comprise any system, device, or apparatus
configured to convert sound incident at such microphone to an
electrical signal that may be processed by a controller, and may
include without limitation an electrostatic microphone, a condenser
microphone, an electret microphone, an analog
microelectromechanical systems (MEMS) microphone, a digital MEMS
microphone, a piezoelectric microphone, a piezo-ceramic microphone,
or dynamic microphone.
[0023] Referring now to FIG. 2, selected circuits within personal
audio device 10, which in other embodiments may be placed in whole
or part in other locations such as one or more headphone assemblies
13, are shown in a block diagram. CODEC IC 20 may include an
analog-to-digital converter (ADC) 21A for receiving the reference
microphone signal and generating a digital representation ref of
the reference microphone signal, an ADC 21B for receiving the error
microphone signal and generating a digital representation err of
the error microphone signal, and an ADC 21C for receiving the near
speech microphone signal and generating a digital representation ns
of the near speech microphone signal. CODEC IC 20 may generate an
output for driving speaker SPKR from an amplifier A1, which may
amplify the output of a digital-to-analog converter (DAC) 23 that
receives the output of a combiner 26. Combiner 26 may combine an
equalized source audio signal generated by adaptive equalization
circuit 40 from audio signals is from internal audio sources 24
and/or downlink speech ds which may be received from radio
frequency (RF) integrated circuit 22, the anti-noise signal
generated by ANC circuit 30, which by convention has the same
polarity as the noise in reference microphone signal ref and is
therefore subtracted by combiner 26, and a portion of near speech
microphone signal ns so that the user of personal audio device 10
may hear his or her own voice in proper relation to downlink speech
ds. Near speech microphone signal ns may also be provided to RF
integrated circuit 22 and may be transmitted as uplink speech to
the service provider via antenna ANT.
[0024] Referring now to FIG. 3, details of ANC circuit 30 are shown
in accordance with embodiments of the present disclosure. Adaptive
filter 32 may receive reference microphone signal ref and under
ideal circumstances, may adapt its transfer function W(z) to be
P(z)/S(z) to generate the anti-noise signal, which may be provided
to an output combiner that combines the anti-noise signal with the
audio to be reproduced by the transducer, as exemplified by
combiner 26 of FIG. 2. The coefficients of adaptive filter 32 may
be controlled by a W coefficient control block 31 that uses a
correlation of signals to determine the response of adaptive filter
32, which generally minimizes the error, in a least-mean squares
sense, between those components of reference microphone signal ref
present in error microphone signal err. The signals compared by W
coefficient control block 31 may be the reference microphone signal
ref as shaped by a copy of an estimate of the response of path S(z)
provided by filter 34B and a playback corrected error, labeled as
"PBCE" in FIG. 3, based at least in part on error microphone signal
err. The playback corrected error may be generated as described in
greater detail below.
[0025] By transforming reference microphone signal ref with a copy
of the estimate of the response of path S(z), response
SE.sub.COPY(z) of filter 34B, and minimizing the difference between
the resultant signal and error microphone signal err, adaptive
filter 32 may adapt to the desired response of P(z)/S(z). In
addition to error microphone signal err, the signal compared to the
output of filter 34B by W coefficient control block 31 may include
an inverted amount of equalized source audio signal (e.g., downlink
audio signal ds and/or internal audio signal ia), that has been
processed by filter response SE(z), of which response
SE.sub.COPY(z) is a copy. By injecting an inverted amount of
equalized source audio signal, adaptive filter 32 may be prevented
from adapting to the relatively large amount of equalized source
audio signal present in error microphone signal err. However, by
transforming that inverted copy of equalized source audio signal
with the estimate of the response of path S(z), the equalized
source audio that is removed from error microphone signal err
should match the expected version of the equalized source audio
signal reproduced at error microphone signal err, because the
electrical and acoustical path of S(z) is the path taken by the
equalized source audio signal to arrive at error microphone E.
Filter 34B may not be an adaptive filter, per se, but may have an
adjustable response that is tuned to match the response of adaptive
filter 34A, so that the response of filter 34B tracks the adapting
of adaptive filter 34A.
[0026] To implement the above, adaptive filter 34A may have
coefficients controlled by SE coefficient control block 33, which
may compare the equalized source audio signal and a playback
corrected error. The playback corrected error may be equal to error
microphone signal err after removal of the equalized source audio
signal (as filtered by filter 34A to represent the expected
playback audio delivered to error microphone E) by a combiner 36.
SE coefficient control block 33 may correlate the actual equalized
source audio signal with the components of the equalized source
audio signal that are present in error microphone signal err.
Adaptive filter 34A may thereby be adapted to generate a secondary
estimate signal from the equalized source audio signal, that when
subtracted from error microphone signal err to generate the
playback corrected error, includes the content of error microphone
signal err that is not due to the equalized source audio
signal.
[0027] Although FIGS. 2 and 3 depict a feedforward ANC system in
which an anti-noise signal is generated from a filtered reference
microphone signal, any other suitable ANC system employing an error
microphone may be used in connection with the methods and systems
disclosed herein. For example, in some embodiments, an ANC circuit
employing feedback ANC, in which anti-noise is generated from a
playback corrected error signal, may be used instead of or in
addition to feedforward ANC, as depicted in FIGS. 2 and 3.
[0028] Referring now to FIG. 4, details of adaptive equalizer
circuit 40 are shown in accordance with embodiments of the present
disclosure. Adaptive equalization filter 42 may receive the source
audio signal (e.g., downlink speech ds and/or internal audio ia)
and under ideal circumstances, may adapt its transfer function
EQ(z) to be Delay/S(z) (wherein Delay is a signal delay added to a
signal by delay element 48, as described in greater detail below)
to generate the equalized source audio signal, which may be
provided to ANC circuit 30 (as described above) and provided to an
output combiner that combines the anti-noise signal with the
equalized source audio signal to be reproduced by the transducer,
as exemplified by combiner 26 of FIG. 2. The coefficients of
adaptive equalization filter 42 may be controlled by an equalizer
coefficient control block 41 that uses a correlation of signals to
determine the response EQ(z) of adaptive equalization filter 42,
which generally minimizes the error, in a least-mean squares sense,
between the delayed source audio signal and the error microphone
signal err, as described in greater detail below.
[0029] To implement the above, adaptive equalization filter 42 may
have coefficients controlled by equalizer coefficient control block
41, which may compare a source audio signal and a delay corrected
error. The source audio signal may include downlink audio signal ds
and/or internal audio signal ia. The delay corrected error may be
equal to error microphone signal err after removal of the source
audio signal (as delayed by a delay block 48) by a combiner 46.
Equalization coefficient control block 41 may correlate the actual
source audio signal with the components of the source audio signal
that are present in error microphone signal err. The signals
compared by equalizer coefficient control block 41 may be the
source audio signal as shaped by a copy of an estimate of the
response of path S(z) provided by filter 34C and a delay corrected
error, based at least in part on error microphone signal err.
[0030] In some embodiments, adaptive equalization filter 42 may
comprise a shelving filter, as is known in the art. In such
embodiments, at least one of a pole frequency and a zero frequency
of the shelving filter may be variable based on the error
microphone signal.
[0031] As mentioned above, in addition to error microphone signal
err, the signal compared to the output of filter 34C by equalizer
coefficient control block 41 may include a delayed amount source
audio signal (e.g., downlink audio signal ds and/or internal audio
signal ia), that has been delayed by delay block 48. By delaying
the source audio signal by at least the delay of the secondary path
represented by S(z), the system formed by adaptive equalization
circuit 40 may operate as a causal system.
[0032] In some embodiments, a noise injection portion 50 may inject
noise into each side of equalizer coefficient control block 41, as
shown in FIG. 4. For example, noise injection portion 50 may inject
an x-side injected noise signal into the filtered source audio
signal generated by filter 34C (e.g., by a combiner which is not
explicitly shown) and an e-side injected noise signal into the
delay corrected error (e.g., by combiner 46 or another combiner
which is not explicitly shown).
[0033] Referring now to FIG. 5, details of a noise injection
portion 50, which may be present in some embodiments of adaptive
equalizer circuit 40 in or are shown in accordance with embodiments
of the present disclosure. Noise injection portion 50 may include a
white noise source 54 for generating white noise (e.g., an audio
signal with a constant amplitude across all frequencies of
interest, such as those frequencies within the range of human
hearing). A frequency shaping filter 56 may generate the x-side
injected noise signal by filtering the white noise signal, wherein
a response of the frequency shaping filter is shaped by frequency
shaping filter coefficient control block 58 in conformity with the
playback corrected error, response SE(z) of filter 34A, or other
suitable signal or response. In some embodiments, coefficient
control block 58 may implement an adaptive linear prediction
coefficient system which estimates a frequency spectrum of the
playback corrected error, response SE(z) of filter 34A, or other
suitable signal or response received by noise injection portion 50.
Accordingly, the noise signal generated by frequency shaping filter
56 may comprise the white noise signal filtered such that the white
noise signal is attenuated or eliminated in those frequencies
within the frequency spectrum of the playback corrected error, such
that the output of frequency shaping filter 56 has a frequency
spectrum with greater magnitude content at frequencies in which the
playback corrected error, response SE(z) of filter 34A, or other
suitable signal or response received by noise injection portion 50
is at or is substantially near zero. In these and other
embodiments, noise injection portion 50 may include an adaptive
equalizer filter 42B, which may be a copy of adaptive equalization
filter 42, wherein adaptive equalizer filter 42B applies its
response EQ.sub.COPY(z) to the x-side injection noise, in order to
generate the e-side injection noise signal. The injected noise
signals may serve to bias, to below a predetermined maximum, a
magnitude of the response of adaptive equalization filter 42
corresponding to a frequency in which the response of secondary
path estimate filter 34C is substantially zero.
[0034] In addition to or alternatively to the noise injection
described above, other approaches may be used in order to limit
magnitudes of the response of adaptive equalization filter 42 at
frequencies corresponding to nulls in the response SE(z) below a
predetermined acceptable level. For example, in some embodiments, a
number of coefficients of adaptive equalizer filter 42 and
equalizer coefficient control block 41 may be selected in order to
limit magnitudes of the response of adaptive equalization filter 42
at frequencies corresponding to nulls in the response SE(z) below a
predetermined acceptable level.
[0035] In these and other embodiments, the response of adaptive
equalizer filter 42 may be disabled from adapting when conditions
are present that may hinder the ability of adaptive equalizer
filter 42 to converge or adapt. For example, the response of
adaptive equalizer filter 42 may be disabled from adapting when the
spectral density of the source audio signal is lesser than a
minimum spectral density. As another example, the response of
adaptive equalizer filter 42 may be disabled from adapting when a
transducer has been removed from a proximity of an ear of a
listener (which may be determined as described in U.S. patent
application Ser. No. 13/844,602 filed Mar. 15, 2013, entitled
"Monitoring of Speaker Impedance to Detect Pressure Applied Between
Mobile Device in Ear," as described in U.S. patent application Ser.
No. 13/310,380 filed Dec. 2, 2011, entitled "Ear-Coupling Detection
and Adjustment of Adaptive Response in Noise-Cancelling in Personal
Audio Devices," or as otherwise known in the art). As an additional
example, the response of adaptive equalizer filter 42 may be
disabled from adapting when "clipping" may occur, as indicated by a
magnitude of the audio output signal driving a transducer being
within a predetermined threshold of a magnitude of a power supply
for driving the output audio signal. As a further example, the
response of adaptive equalizer filter 42 may be disabled from
adapting when a physical displacement of a transducer is such that
its displacement as a function of the output audio signal driving
the transducer is substantially nonlinear.
[0036] In some embodiments, the sequencing of adaptation of
response SE(z) of filter 34A and response EQ(z) of adaptive
equalization filter 42 may be configured to ensure stability of
adaptation of response SE(z) and response EQ(z). For example, in
such embodiments, CODEC IC 20 may be configured to train response
SE(z) prior to training of response EQ(z), as response EQ(z) relies
on response SE.sub.COPY(z) for stability. After both responses
SE(z) and EQ(z) have been trained, training may alternate between
the responses. As another example, CODEC IC 20 may be configured to
such that response EQ(z) trains only while response SE(z) is
training, again because response EQ(z) relies on response
SE.sub.COPY(z) for stability. As a further example, CODEC IC 20 may
be configured such that response EQ(z) adapts at a slower rate than
response SE(z).
[0037] This disclosure encompasses all changes, substitutions,
variations, alterations, and modifications to the example
embodiments herein that a person having ordinary skill in the art
would comprehend. Similarly, where appropriate, the appended claims
encompass all changes, substitutions, variations, alterations, and
modifications to the example embodiments herein that a person
having ordinary skill in the art would comprehend. Moreover,
reference in the appended claims to an apparatus or system or a
component of an apparatus or system being adapted to, arranged to,
capable of, configured to, enabled to, operable to, or operative to
perform a particular function encompasses that apparatus, system,
or component, whether or not it or that particular function is
activated, turned on, or unlocked, as long as that apparatus,
system, or component is so adapted, arranged, capable, configured,
enabled, operable, or operative.
[0038] All examples and conditional language recited herein are
intended for pedagogical objects to aid the reader in understanding
the invention and the concepts contributed by the inventor to
furthering the art, and are construed as being without limitation
to such specifically recited examples and conditions. Although
embodiments of the present inventions have been described in
detail, it should be understood that various changes,
substitutions, and alterations could be made hereto without
departing from the spirit and scope of the disclosure.
* * * * *